The mouth-to-ear latency or delay of data packet streams in rich-media conferences often determines the usability of interaction for the participants. To simplify measurement, latencies are often computed as round-trip latencies; that is, double the one-way or unidirectional mouth-to-ear latency. By way of example, round-trip audio latencies in excess of 100 milliseconds degrade conference quality, since the delay between one participant speaking and a next participant speaking causes interruptions and overlap to occur. A conference with round-trip audio latencies of 300 milliseconds suffers such severely degraded audio quality that the conference participants are usually dissatisfied with the experience.
Round-trip delay of media streams in conferencing systems is a function of many factors, including packet formation delay, network latency, jitter, and other computation phenomena involved in rendering media. End users today have no easy way of determining point-to-point or round-trip latency for a given (i.e., arbitrary) conferencing or telephony system. Some conferencing systems have built-in latency measurement tools; however, those tool are generally incapable of measuring the overall delay (i.e., from the mouth speaking into a microphone on an endpoint device, through the conferencing bridge/mixer/server, to the ear listening to a loudspeaker on another endpoint). Furthermore, such systems do not always work with third-party endpoint devices. These systems also fail to measure delays in the case of two or more interworking conferencing systems.